In addition to traditional speech services, mobile communication systems, such as the GSM system (Global System for Mobile communication), typically also provide non-speech services, for example transmission of texts, images, facsimile or computer files. These non-speech services are usually referred to as data services. The basic requirements for successful transmission of data services differ remarkably from those of speech services. Since errors are not allowed in data transmission, data services must be handled differently within a mobile communication network, i.e. on a transmission path from a mobile station (MS) to a mobile communication switching centre (MSC). Furthermore, data transmission between a mobile communication network, such as the GSM, and an external communication network, for example an ISDN (Integrated Services Digital Network), a PSTN (Public Switched Telephone Network) or a PDN (Public Data Network), requires a specific adaptation function called an ‘Interworking Function’ (IWF).
In the GSM system, the transmission path from the MS to the MSC goes through a radio interface Um to a base transceiver station (BTS) and via an Abis interface further to a base station controller (BSC) and from there via an A interface to a mobile switching centre (MSC). Somewhere between the BTS and the MSC, either on the Abis interface or on the A interface, there is a transcoder/rate adaptor unit (TRAU), the task of which is to raise and adapt the transmission rate of the bit flow arriving from the MS via the BTS up to 64 kbit/s. Physically, the TRAU can be located in connection with the BTS, the BSC or the MSC. However, the TRAU cannot be located functionally inside the MSC, but logically, it must always be situated before the MSC. Therefore, the speed of the incoming information flow via the A interface towards the MSC is always 64 kbit/s, regardless of whether the information flow consists of speech or data.
Data rate adaptation problems similar to those previously solved in ISDN data rate adaptation towards external networks arose in the design of the GSM system. As a result, a similar approach, based on the V.110 protocol, was adopted in the GSM system to handle the interface between the TRAU via the MSC to the IWF (referred to as TRAU/IWF). V.110 is an ITU (International Telecommunication Union) standard technique for data rate adaptation originally designed for adaptors between a PSTN and an ISDN. The V.110 protocol utilizes TDM frames together with end-to-end control signals for modems, and it supports both synchronous and asynchronous data . There are no error detection or correction functions. Thus, data is transmitted between the TRAU and the IWF in V.110 frames.
The IWF comprises means for relaying the transmission further to external networks. The relaying means comprise ISDN means for redirecting and reformatting the V.110 frames to the ISDN network almost as such, and a modem for modulating the data transmission towards a PSTN network or the like. Thus, the relaying means used are selected on the basis of the type of the external network.
The latest development in IP-based (Internet Protocol) networks, which are typically designed for packet data transmission, has also aroused interest in the use of these networks for the transmission of voice calls. IP-based networks are becoming very common and they offer lowcost connections or even free connections, for example in local area networks (LAN). IP connections have been introduced in inter-MSC connections in mobile networks, too. A general recommendation, Voice over IP (VoIP), is used to define, inter alia, hardware compatibility for a connection, quality of service and routing of calls. For inter-MSC IP connections, a VoIP gateway is provided on both sides of the IP connection to compress the 64 kbit/s speech transmission from the MSC. The speech is compressed at 8 kbit/s, for example, to save capacity in the IP network, and on the opposite side the other VoIP gateway decompresses the speech back to 64 kbit/s to be delivered to the respective MSC.
The problem with the arrangement described above is that the current VoIP gateways are only applicable in respect of speech and fax calls. That is, if a data call using V.110 frames is to be delivered on an inter-MSC IP connection, the VoIP gateway assumes the arriving information flow to be a speech call and tries to compress it with a speech codec. The speech codec cannot distinguish the pulse-code modulated (PCM) speech frames and V.110 data frames, wherefore the VoIP gateway compresses the data call into illegible form, which cannot be decompressed any longer.